Yes, I know that MP3Gain is lossless. No, I am not talking about how MP3Gain is limited to 1.5dB steps (though this is another reason).What I am thinking is this:

Psychoacoustic models tend to encode louder signals with more bits, right? This is why remasters (compressed) tend to use higher bitrates than older versions. Correct?

If we are then encoding a VERY LOUD album with LAME -aps for instance, then normalizing it down say 10.5dB (for a really bad one), aren't we wasting bits? Wouldn't we be able to save some bits by normalizing it down to 89dB FIRST, then encoding it?

I know wavegain is NOT LOSSLESS, but I am (incorrectly) assuming that this is just a theoretical lossiness, not a perceptible one. If this is the case, couldn't we argue that wavegained tracks will be of lower bitrates while still maintaining transparency (the goal, afterall, of -aps)?

The discussion below basically concludes that running a wavegain analysis, then applying the recommended scalefactor to lame (via the --scale switch) will save bits due to high-frequency bloat inherent in the mp3 format. Lots of people have since chosen to use this method over mp3gain, and in fact it can be automated now from within EAC using Wack, or my own Omni Encoder. Read on!

I don't think that going to a greater bitdepth, or even to floating-point calculations, alters the problem in principle. If we carried out the entire wavgain process, for example, from a floating-point input all the way down to floating-point silence, we have still lost all the information in the original signal; the asymptote is still complete loss.

I agree that in practice the amount of loss is likely to be so small as to be imperceptible, and that higher bitdepths will eliminate noise that would be introduced by dithering otherwise. But the basic problem is still there.

I completely agree with you.You explain it in an earlier post too (same thread, page 1).I am not sure how the code works at all, I haven't even looked at it to tell you the truth, but I think mrosscook's point is valid from a logical point of view, assuming that the floor is at a constant value and that values below that floor are dropped. If we apply a negative gain, everything in that file will drop by that much. Keep in mind that before the gain was applied there was some sound info that was just above the floor limit. If all that is true, then there should be some sound info that drops below the floor limit if we apply a negative gain (ie is dropped from the file), meaning there is less sound info after the gain process to encode. It also means that if you take the same track, and apply two different gains to it, one being more negative than the other, the resulting files should have two different sizes.